Added buildbot benchmarking in iSAC and APM into Android platform build.
Review URL: https://webrtc-codereview.appspot.com/964022
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@3247 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/modules/audio_coding/codecs/isac/fix/source/Android.mk b/modules/audio_coding/codecs/isac/fix/source/Android.mk
index e7745ea..888fbd8 100644
--- a/modules/audio_coding/codecs/isac/fix/source/Android.mk
+++ b/modules/audio_coding/codecs/isac/fix/source/Android.mk
@@ -119,7 +119,7 @@
LOCAL_MODULE_TAGS := tests
LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= ../test/kenny.c
+LOCAL_SRC_FILES:= ../test/kenny.cc
# Flags passed to both C and C++ files.
LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
@@ -131,7 +131,8 @@
LOCAL_STATIC_LIBRARIES := \
libwebrtc_isacfix \
libwebrtc_spl \
- libwebrtc_system_wrappers
+ libwebrtc_system_wrappers \
+ libwebrtc_test_support
ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
LOCAL_STATIC_LIBRARIES += \
diff --git a/modules/audio_coding/codecs/isac/fix/test/kenny.c b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
similarity index 97%
rename from modules/audio_coding/codecs/isac/fix/test/kenny.c
rename to modules/audio_coding/codecs/isac/fix/test/kenny.cc
index b7ca694..d95dea9 100644
--- a/modules/audio_coding/codecs/isac/fix/test/kenny.c
+++ b/modules/audio_coding/codecs/isac/fix/test/kenny.cc
@@ -8,16 +8,17 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-/* kenny.c - Main function for the iSAC coder */
-
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <time.h>
#include <ctype.h>
-#include "isacfix.h"
+#include "webrtc/modules/audio_coding/codecs/isac/fix/interface/isacfix.h"
+#include "webrtc/test/testsupport/perf_test.h"
+// TODO(kma): Clean up the code and change benchmarking the whole codec to
+// separate encoder and decoder.
/* Defines */
#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
@@ -170,7 +171,7 @@
" in adaptive mode.\n\n");
printf("[-FL num] :Set (initial) frame length in msec. Valid length"
" are 30 and 60 msec.\n\n");
- printf("[-FIXED_FL] :Frame length will be fixed to initial value.\n\n");
+ printf("[-FIXED_FL] :Frame length to be fixed to initial value.\n\n");
printf("[-MAX num] :Set the limit for the payload size of iSAC"
" in bytes. \n");
printf(" Minimum 100, maximum 400.\n\n");
@@ -374,7 +375,8 @@
sscanf(argv[CodingMode+1], "%s", bottleneck_file);
f_bn = fopen(bottleneck_file, "rb");
if (f_bn == NULL) {
- printf("No value provided for BottleNeck and cannot read file %s\n", bottleneck_file);
+ printf("No value provided for BottleNeck and cannot read file %s\n",
+ bottleneck_file);
exit(0);
} else {
int aux_var;
@@ -565,8 +567,8 @@
shortdata,
(WebRtc_Word16*)streamdata);
- /* If packet is ready, and CE testing, call the different API functions
- from the internal API. */
+ /* If packet is ready, and CE testing, call the different API
+ functions from the internal API. */
if (stream_len>0) {
if (testCE == 1) {
err = WebRtcIsacfix_ReadBwIndex((WebRtc_Word16*)streamdata, &bwe);
@@ -808,6 +810,10 @@
runtime, (100*runtime/length_file));
printf("\n\n_______________________________________________\n");
+ // Record the results with Perf test tools.
+ webrtc::test::PrintResult("time_per_10ms_frame", "", "isac",
+ (runtime * 10000) / length_file, "us", false);
+
fclose(inp);
fclose(outp);
fclose(outbits);
diff --git a/modules/audio_coding/codecs/isac/isacfix_test.gypi b/modules/audio_coding/codecs/isac/isacfix_test.gypi
index e1e1311..ce2f46c 100644
--- a/modules/audio_coding/codecs/isac/isacfix_test.gypi
+++ b/modules/audio_coding/codecs/isac/isacfix_test.gypi
@@ -14,13 +14,14 @@
'type': 'executable',
'dependencies': [
'iSACFix',
+ '<(webrtc_root)/test/test.gyp:test_support',
],
'include_dirs': [
'./fix/test',
'./fix/interface',
],
'sources': [
- './fix/test/kenny.c',
+ './fix/test/kenny.cc',
],
},
],
diff --git a/modules/audio_processing/Android.mk b/modules/audio_processing/Android.mk
index 201b23a..7e87751 100644
--- a/modules/audio_processing/Android.mk
+++ b/modules/audio_processing/Android.mk
@@ -89,6 +89,7 @@
LOCAL_STATIC_LIBRARIES := \
libgtest \
+ libwebrtc_test_support \
libprotobuf-cpp-2.3.0-lite
LOCAL_SHARED_LIBRARIES := \
@@ -113,8 +114,7 @@
LOCAL_CPP_EXTENSION := .cc
LOCAL_SRC_FILES:= \
$(call all-proto-files-under, test) \
- test/unit_test.cc \
- ../../test/testsupport/fileutils.cc
+ test/unit_test.cc
# Flags passed to both C and C++ files.
LOCAL_CFLAGS := \
@@ -136,6 +136,7 @@
LOCAL_STATIC_LIBRARIES := \
libgtest \
+ libwebrtc_test_support \
libprotobuf-cpp-2.3.0-lite
LOCAL_SHARED_LIBRARIES := \
diff --git a/modules/audio_processing/audio_processing_tests.gypi b/modules/audio_processing/audio_processing_tests.gypi
index 0d5bfac..323a457 100644
--- a/modules/audio_processing/audio_processing_tests.gypi
+++ b/modules/audio_processing/audio_processing_tests.gypi
@@ -58,8 +58,9 @@
'dependencies': [
'audio_processing',
'audioproc_debug_proto',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
'<(DEPTH)/testing/gtest.gyp:gtest',
+ '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
+ '<(webrtc_root)/test/test.gyp:test_support',
],
'sources': [ 'test/process_test.cc', ],
},
diff --git a/modules/audio_processing/test/process_test.cc b/modules/audio_processing/test/process_test.cc
index e68d1e9..70c4b15 100644
--- a/modules/audio_processing/test/process_test.cc
+++ b/modules/audio_processing/test/process_test.cc
@@ -19,11 +19,12 @@
#include "gtest/gtest.h"
-#include "audio_processing.h"
-#include "cpu_features_wrapper.h"
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "tick_util.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
+#include "webrtc/test/testsupport/perf_test.h"
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_processing/debug.pb.h"
#else
@@ -1032,6 +1033,9 @@
(exec_time * 1.0) / primary_count,
(max_time_us + max_time_reverse_us) / 1000.0,
(min_time_us + min_time_reverse_us) / 1000.0);
+ // Record the results with Perf test tools.
+ webrtc::test::PrintResult("time_per_10ms_frame", "", "audioproc",
+ (exec_time * 1000) / primary_count, "us", false);
} else {
printf("Warning: no capture frames\n");
}
diff --git a/test/Android.mk b/test/Android.mk
new file mode 100644
index 0000000..20af6f9
--- /dev/null
+++ b/test/Android.mk
@@ -0,0 +1,39 @@
+
+# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
+#
+# Use of this source code is governed by a BSD-style license
+# that can be found in the LICENSE file in the root of the source
+# tree. An additional intellectual property rights grant can be found
+# in the file PATENTS. All contributing project authors may
+# be found in the AUTHORS file in the root of the source tree.
+
+LOCAL_PATH := $(call my-dir)
+
+include $(CLEAR_VARS)
+include $(LOCAL_PATH)/../../android-webrtc.mk
+
+LOCAL_ARM_MODE := arm
+LOCAL_MODULE:= libwebrtc_test_support
+LOCAL_MODULE_TAGS := optional
+LOCAL_CPP_EXTENSION := .cc
+LOCAL_SRC_FILES:= \
+ testsupport/fileutils.cc \
+ testsupport/perf_test.cc
+
+# Flags passed to both C and C++ files.
+LOCAL_CFLAGS := \
+ $(MY_WEBRTC_COMMON_DEFS)
+
+LOCAL_C_INCLUDES := \
+ external/gtest/include \
+ external/webrtc \
+ external/webrtc/webrtc
+
+LOCAL_STATIC_LIBRARIES := \
+ libgtest
+
+ifndef NDK_ROOT
+include external/stlport/libstlport.mk
+endif
+include $(BUILD_STATIC_LIBRARY)
+
